1. Field of the Invention
The present invention relates generally to dynamically changing compression techniques used for transmission of voice information over a packet switching environment and for transmission of fax information on a telephone line, in real-time, the telephone line being also used for transmission of voice information over a packet switching network environment and particularly for loading an overlay compression layer during a telephone conversation upon user initiation or automatically.
2. Description of the Prior Art
Prior art systems utilize various ways of transmitting voice information. One such way is shown in FIG. 1 wherein a telephone system 10 is shown to include a telephone device 12 coupled to a local telephone exchange, i.e. California AT&T exchange 14 (assuming the telephone device 12 is located in California). The latter is coupled through a PSTN 16 (Public Switching Telephone Network) to another exchange, namely, a New York telephone exchange 18 and finally, the exchange 18 is coupled to communicate voice information to a telephone device 20. This setup has been traditionally employed for communicating voice telephone calls between remotely located users of telephones, such as users of the telephone devices 12 and 20 (not shown in FIG. 1).
In operation, a user of the telephone device 12, located in California, might place a telephone call to a user of the telephone device 20, which may be located in New York. Once the call is initiated, a telephone transmission line establishes a connection between the telephone device 12 and the exchange 14. At the exchange 14, the telephone line might be multiplexed with a number of other physical lines and transmitted over the PSTN 16, which is comprised of switches, to the exchange 18, which is located in New York. The exchange 18 de-multiplexes the telephone line carrying the transmitted call and transfers the de-multiplexed signal to the telephone device 20.
The transmission of information through systems such as that shown in FIG. 1 is costly due to toll charges by telephone companies for placing calls through such systems.
More recently, telephone conversations or modem information has been transmitted using packet switching network environments. An example of such a system is shown in FIG. 2 wherein a prior art telephone system 22 is shown to include a telephone device 24 coupled to an exchange 26, which is in turn coupled to a packet transmission system 36. The packet transmission system 36 is coupled to an exchange 32, which is coupled to a telephone device 34. The packet transmission system 36 is shown to include a California access server 28 and a New York access server 30.
In operation, a telephone call is placed by a user of the telephone device 24 and connected to the exchange 26. In this case, a user of the telephone device is again located in California and therefore the exchange 26 is located locally in California. The exchange 26 couples a multiplexed line with the phone call coupled thereon to the server 28. The server 28 is also locally located, i.e. in California, and operates to converts the voice information, which is received by the server 28 in analog format, to digital format and transforms the same to packets of voice information for transmission through a packet switching network, such as the Internet. The server 28 is a network device that typically includes a router for packetizing and de-packetizing information. Subsequently, the server 30 receives the packetized voice information that was transmitted through the packet switching network and the former de-packetizes (arranges the packets of a telephone call together into a continuous signal) the telephone call information and converts it into analog format for transmission to the exchange 32. The server 30 and the exchange 32 are located remotely from the exchange 26 and server 28. In this example, as the call is being placed between a California user and a New York user, the server 30 and the exchange 32 are located in New York. The server 30 also typically includes a router. The exchange 32 then transmits the voice information, in analog form to the telephone device 34. This type of voice transmission is commonly referred to as voice-over-IP (Internet Protocol). In comparing the prior art systems of FIG. 1 and FIG. 2, it should be noted that the PSTN 16 is essentially replaced with the packet transmission system 36. This difference in the two systems is, however, transparent to users of the telephone devices except that use of the system 22 in FIG. 2 is less expensive than that of the system 10 in FIG. 1. The reason for the cost reduction is that, referring to FIG. 2, the transmission of voice information from the telephone device 24 all the way to the server 28 is through local medium, which generates local phone costs. The same holds true for transmissions of voice information between the telephone device 34 and the server 30, as these are local within New York. Therefore, long distance charges are only incurred through the packet switching network, such as the Internet, which is generally a much less expensive medium of transmission relative to the public switching network.
In FIG. 2, the packets of voice information through the packet switching network, such as the Internet, are transmitted using various encoding techniques. In one encoding technique, packets are compressed by the server (for example, server 28) prior to transmission thereof. Consider the case where each packet is 200 bytes in length. A compression technique, such as one in the list provided hereinbelow, may compress each packet to 25 bytes. The goal is to decrease the size of the packets as much as possible while maintaining signal quality. Smaller packet sizes increase system throughput and therefore capacity as more packets can be routed through the network. Generally, undergoing any type of compression results in compromising voice quality. However, the difference in voice quality is not generally noticed by users listening to the voice transmission due to certain insensitivity associated with the human ear beyond certain frequencies. Although in the example provide hereinabove, a constant packet size, i.e. 200 bytes, is used, in practice, different packet sizes may be transmitted through the network.
A problem that arises with respect to prior art systems similar to FIG. 2 is that only one type of compression algorithm is employed for the duration of a phone call. That is, compression algorithms can not be changed during the phone conversation. The use of only one type of compression technique prevents compensation for variable factors, such as varying packet sizes. Therefore, the need arises for the use of dynamically changing compression techniques, either manually or automatically for transmission of voice-over-IP allowing for adjustments to be made in accordance with varying network sage or bandwidth thereby making optimal use of network capacity and throughput.
Compression and decompression are typically referred to as codec and examples of codecs used for compression/decompression of voice information are: G.711, G.723.1 and G729, etc. Typically, DSP devices perform codec functions. Different codecs offer different advantages and disadvantages. For example, the G.711 codec actually performs no compression thereby leading to increased bandwidth. However, the quality of the voice transmission is as good as the original voice. Codecs are typically located in the router within the servers 28 and 30.
In the system of FIG. 2, fax information, as opposed to voice information, may be transmitted. In this respect, a fax machine is employed in place of the telephone device 24 and similarly, a fax machine is employed in place of the telephone device 34. In fax transmissions, a codec is loaded into the DSP followed by ‘overlay’. ‘overlay’ converts the rate of transmissions of fax signals to the appropriate speed necessary for transmission of fax information over IP. When a fax call is initiated, the router within the server 28 detects a fax tone, which indicates that the information is fax information and accordingly places an ‘overlay’ on top of the already-compressed fax signal. Currently, a user of a telephone/fax device cannot transmit fax information over the same line that the user is utilizing for transmission of voice information. This serves as an inconvenience to the user as the user must place two calls, one for conducting a voice transmission and another for transmission of fax information. Moreover, if the user must fax a document urgently, while on the phone, the user must hang up the voice call, fax the document and then make a third call to resume the voice call. Accordingly, the need arises to transmit fax information, in real-time, over a telephone line that is being used to transmit voice information in a packet switching network environment.
For the foregoing reasons, the need arises for employing dynamically varying compression techniques, either manually or automatically, in packet switching network environments allowing for adjustments to be made by varying the compression technique in accordance to the usage of the network. A further need arises for transmission of fax information, on-the-fly (or in real-time), over the same telephone line that is being used to transmit voice information in a packet switching network.